| /external/webrtc/webrtc/test/ |
| H A D | layer_filtering_transport.cc | 59 RTC_DCHECK_GT(length, header.headerLength);
61 RTC_DCHECK_GT(payload_length, header.paddingLength);
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| H A D | frame_generator.cc | 150 RTC_DCHECK_GT(num_frames_, 0u);
155 RTC_DCHECK_GT(scroll_time_ms + pause_time_ms, 0);
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| /external/webrtc/webrtc/common_video/ |
| H A D | video_frame_buffer.cc | 45 RTC_DCHECK_GT(width, 0);
46 RTC_DCHECK_GT(height, 0);
112 RTC_DCHECK_GT(width, 0);
113 RTC_DCHECK_GT(height, 0);
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| H A D | video_frame.cc | 65 RTC_DCHECK_GT(width, 0);
66 RTC_DCHECK_GT(height, 0);
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| /external/webrtc/webrtc/modules/video_coding/codecs/vp9/ |
| H A D | screenshare_layers.cc | 21 RTC_DCHECK_GT(num_layers, 0);
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| H A D | vp9_frame_buffer_pool.cc | 56 RTC_DCHECK_GT(min_size, 0u);
|
| /external/webrtc/webrtc/modules/audio_processing/beamformer/ |
| H A D | array_util.cc | 61 RTC_DCHECK_GT(array_geometry.size(), 1u);
76 RTC_DCHECK_GT(array_geometry.size(), 1u);
|
| H A D | nonlinear_beamformer.cc | 244 RTC_DCHECK_GT(low_mean_start_bin_, 0U);
550 RTC_DCHECK_GT(last, first);
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| /external/webrtc/webrtc/base/ |
| H A D | checks.h | 174 #define RTC_DCHECK_GT(v1, v2) RTC_CHECK_GT(v1, v2) macro
183 #define RTC_DCHECK_GT(v1, v2) RTC_EAT_STREAM_PARAMETERS((v1) > (v2))
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| H A D | filerotatingstream.cc | 40 RTC_DCHECK_GT(max_file_size, 0u);
41 RTC_DCHECK_GT(num_files, 1u);
|
| /external/webrtc/webrtc/common_audio/include/ |
| H A D | audio_util.h | 157 RTC_DCHECK_GT(num_channels, 0);
158 RTC_DCHECK_GT(num_frames, 0u);
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| /external/webrtc/webrtc/video/ |
| H A D | receive_statistics_proxy.cc | 167 RTC_DCHECK_GT(width, 0);
168 RTC_DCHECK_GT(height, 0);
|
| H A D | video_send_stream.cc | 393 RTC_DCHECK_GT(streams[i].width, 0u);
394 RTC_DCHECK_GT(streams[i].height, 0u);
395 RTC_DCHECK_GT(streams[i].max_framerate, 0);
428 RTC_DCHECK_GT(streams[0].max_framerate, 0);
|
| /external/webrtc/webrtc/modules/audio_coding/codecs/opus/ |
| H A D | audio_encoder_opus_unittest.cc | 108 RTC_DCHECK_GT(n, 1u);
|
| /external/webrtc/webrtc/modules/audio_device/android/ |
| H A D | audio_device_template.h | 391 RTC_DCHECK_GT(delay_ms, 0);
398 RTC_DCHECK_GT(delay_ms, 0);
|
| H A D | audio_record_jni.cc | 193 RTC_DCHECK_GT(total_delay_in_milliseconds_, 0);
|
| /external/webrtc/webrtc/modules/rtp_rtcp/source/ |
| H A D | rtp_format_vp9.cc | 101 RTC_DCHECK_GT(hdr.num_ref_pics, 0U);
130 RTC_DCHECK_GT(hdr.num_spatial_layers, 0U);
271 RTC_DCHECK_GT(vp9.num_spatial_layers, 0U);
|
| /external/webrtc/webrtc/modules/bitrate_controller/ |
| H A D | send_side_bandwidth_estimation.cc | 69 RTC_DCHECK_GT(bitrate, 0);
|
| /external/webrtc/webrtc/modules/audio_processing/intelligibility/ |
| H A D | intelligibility_enhancer.cc | 360 RTC_DCHECK_GT(freqs_, 0u);
|
| /external/webrtc/webrtc/modules/pacing/ |
| H A D | paced_sender.cc | 319 RTC_DCHECK_GT(max_bitrate_kbps_, 0);
|
| /external/webrtc/webrtc/p2p/stunprober/ |
| H A D | stunprober.cc | 501 RTC_DCHECK_GT(kv.second, 0);
|
| /external/webrtc/webrtc/call/ |
| H A D | rampup_tests.cc | 222 RTC_DCHECK_GT(expected_bitrate_bps_, 0);
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| H A D | call.cc | 528 RTC_DCHECK_GT(bitrate_config.max_bitrate_bps, 0);
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| /external/webrtc/webrtc/modules/audio_coding/acm2/ |
| H A D | audio_coding_module_impl.cc | 171 RTC_DCHECK_GT(encode_buffer_.size(), 0u);
|
| /external/webrtc/webrtc/voice_engine/ |
| H A D | voe_base_impl.cc | 590 RTC_DCHECK_GT(1024u, versionString.size() + 1);
|